forked from webrtc/samples
-
Notifications
You must be signed in to change notification settings - Fork 0
/
index.html
223 lines (155 loc) · 9.93 KB
/
index.html
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
<!DOCTYPE html>
<!--
* Copyright (c) 2015 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree.
-->
<html>
<head>
<meta charset="utf-8">
<meta name="description" content="WebRTC Javascript code samples">
<meta name="viewport" content="width=device-width, user-scalable=yes, initial-scale=1, maximum-scale=1">
<meta itemprop="description" content="Client-side WebRTC code samples">
<meta itemprop="image" content="src/images/webrtc-icon-192x192.png">
<meta itemprop="name" content="WebRTC code samples">
<meta name="mobile-web-app-capable" content="yes">
<meta id="theme-color" name="theme-color" content="#ffffff">
<base target="_blank">
<title>WebRTC samples</title>
<link rel="icon" sizes="192x192" href="src/images/webrtc-icon-192x192.png">
<link href="https://fonts.googleapis.com/css?family=Roboto:300,400,500,700" rel="stylesheet" type="text/css">
<link rel="stylesheet" href="src/css/main.css"/>
<style>
h2 {
font-size: 1.5em;
font-weight: 500;
}
h3 {
border-top: none;
}
section {
border-bottom: 1px solid #eee;
margin: 0 0 1.5em 0;
padding: 0 0 1.5em 0;
}
section:last-child {
border-bottom: none;
margin: 0;
padding: 0;
}
</style>
</head>
<body>
<div id="container">
<h1>WebRTC samples</h1>
<section>
<p>
This is a collection of small samples demonstrating various parts of the <a
href="https://developer.mozilla.org/en-US/docs/Web/API/WebRTC_API">WebRTC APIs</a>. The code for all
samples are available in the <a href="https://github.com/webrtc/samples">GitHub repository</a>.
</p>
<p>Most of the samples use <a href="https://github.com/webrtc/adapter">adapter.js</a>, a shim to insulate apps
from spec changes and prefix differences.</p>
<p><a href="https://webrtc.org/getting-started/testing" title="Command-line flags for WebRTC testing">https://webrtc.org/getting-started/testing</a>
lists command line flags useful for development and testing with Chrome.</p>
<p>Patches and issues welcome! See <a href="https://github.com/webrtc/samples/blob/gh-pages/CONTRIBUTING.md">CONTRIBUTING.md</a>
for instructions.</p>
<p class="warning"><strong>Warning:</strong> It is highly recommended to use headphones when testing these
samples, as it will otherwise risk loud audio feedback on your system.</p>
</section>
<section>
<h2 id="getusermedia"><a href="https://developer.mozilla.org/en-US/docs/Web/API/MediaDevices/getUserMedia">getUserMedia():</a>
</h2>
<p class="description">Access media devices</p>
<ul>
<li><a href="src/content/getusermedia/gum/">Basic getUserMedia demo</a></li>
<li><a href="src/content/getusermedia/canvas/">Use getUserMedia with canvas</a></li>
<li><a href="src/content/getusermedia/filter/">Use getUserMedia with canvas and CSS filters</a></li>
<li><a href="src/content/getusermedia/resolution/">Choose camera resolution</a></li>
<li><a href="src/content/getusermedia/audio/">Audio-only getUserMedia() output to local audio element</a>
</li>
<li><a href="src/content/getusermedia/volume/">Audio-only getUserMedia() displaying volume</a></li>
<li><a href="src/content/getusermedia/record/">Record stream</a></li>
<li><a href="src/content/getusermedia/getdisplaymedia/">Screensharing with getDisplayMedia</a></li>
<li><a href="src/content/getusermedia/pan-tilt-zoom/">Control camera pan, tilt, and zoom</a></li>
<li><a href="src/content/getusermedia/exposure/">Control exposure</a></li>
</ul>
<h2 id="devices">Devices:</h2>
<p class="description">Query media devices</p>
<ul>
<li><a href="src/content/devices/input-output/">Choose camera, microphone and speaker</a></li>
<li><a href="src/content/devices/multi/">Choose media source and audio output</a></li>
</ul>
<h2 id="capture">Stream capture:</h2>
<p class="description">Stream from canvas or video elements</p>
<ul>
<li><a href="src/content/capture/video-video/">Stream from a video element to a video element</a></li>
<li><a href="src/content/capture/video-pc/">Stream from a video element to a peer connection</a></li>
<li><a href="src/content/capture/canvas-video/">Stream from a canvas element to a video element</a></li>
<li><a href="src/content/capture/canvas-pc/">Stream from a canvas element to a peer connection</a></li>
<li><a href="src/content/capture/canvas-record/">Record a stream from a canvas element</a></li>
<li><a href="src/content/capture/video-contenthint/">Guiding video encoding with content hints</a></li>
</ul>
<h2 id="peerconnection"><a href="https://developer.mozilla.org/en-US/docs/Web/API/RTCPeerConnection">RTCPeerConnection:</a>
</h2>
<p class="description">Controlling peer connectivity</p>
<ul>
<li><a href="src/content/peerconnection/pc1/">Basic peer connection demo in a single tab</a></li>
<li><a href="src/content/peerconnection/channel/">Basic peer connection demo between two tabs</a></li>
<li><a href="src/content/peerconnection/perfect-negotiation/">Peer connection using Perfect Negotiation</a></li>
<li><a href="src/content/peerconnection/audio/">Audio-only peer connection demo</a></li>
<li><a href="src/content/peerconnection/bandwidth/">Change bandwidth on the fly</a></li>
<li><a href="src/content/peerconnection/change-codecs/">Change codecs before the call</a></li>
<li><a href="src/content/peerconnection/upgrade/">Upgrade a call and turn video on</a></li>
<li><a href="src/content/peerconnection/multiple/">Multiple peer connections at once</a></li>
<li><a href="src/content/peerconnection/multiple-relay/">Forward the output of one PC into another</a></li>
<li><a href="src/content/peerconnection/munge-sdp/">Munge SDP parameters</a></li>
<li><a href="src/content/peerconnection/pr-answer/">Use pranswer when setting up a peer connection</a></li>
<li><a href="src/content/peerconnection/constraints/">Constraints and stats</a></li>
<li><a href="src/content/peerconnection/old-new-stats/">More constraints and stats</a></li>
<li><a href="src/content/peerconnection/per-frame-callback/">RTCPeerConnection and requestVideoFrameCallback()</a></li>
<li><a href="src/content/peerconnection/create-offer/">Display createOffer output for various scenarios</a>
</li>
<li><a href="src/content/peerconnection/dtmf/">Use RTCDTMFSender</a></li>
<li><a href="src/content/peerconnection/states/">Display peer connection states</a></li>
<li><a href="src/content/peerconnection/trickle-ice/">ICE candidate gathering from STUN/TURN servers</a>
</li>
<li><a href="src/content/peerconnection/restart-ice/">Do an ICE restart</a></li>
<li><a href="src/content/peerconnection/webaudio-input/">Web Audio output as input to peer connection</a>
</li>
<li><a href="src/content/peerconnection/webaudio-output/">Peer connection as input to Web Audio</a></li>
<li><a href="src/content/peerconnection/negotiate-timing/">Measure how long renegotiation takes</a></li>
<li><a href="src/content/extensions/svc/">Choose scalablilityMode before call - Scalable Video Coding (SVC) Extension </a></li>
</ul>
<h2 id="datachannel"><a
href="https://developer.mozilla.org/en-US/docs/Web/API/RTCDataChannel">RTCDataChannel:</a></h2>
<p class="description">Send arbitrary data over peer connections</p>
<ul>
<li><a href="src/content/datachannel/basic/">Transmit text</a></li>
<li><a href="src/content/datachannel/filetransfer/">Transfer a file</a></li>
<li><a href="src/content/datachannel/datatransfer/">Transfer data</a></li>
<li><a href="src/content/datachannel/channel/">Basic datachannel demo between two tabs</a></li>
<li><a href="src/content/datachannel/messaging/">Messaging</a></li>
</ul>
<h2 id="videoChat">Video chat:</h2>
<p class="description">Full featured WebRTC application</p>
<ul>
<li><a href="https://github.com/webrtc/apprtc/">AppRTC video chat client</a> that you can run out of a Docker image</li>
</ul>
<h2 id="capture">Insertable Streams:</h2>
<p class="description">API for processing media</p>
<ul>
<li><a href="src/content/insertable-streams/endtoend-encryption">End to end encryption using WebRTC Insertable Streams</a></li> (Experimental)
<li><a href="src/content/insertable-streams/video-analyzer">Video analyzer using WebRTC Insertable Streams</a></li> (Experimental)
<li><a href="src/content/insertable-streams/video-processing">Video processing using MediaStream Insertable Streams</a></li> (Experimental)
<li><a href="src/content/insertable-streams/audio-processing">Audio processing using MediaStream Insertable Streams</a></li> (Experimental)
<li><a href="src/content/insertable-streams/video-crop">Video cropping using MediaStream Insertable Streams in a Worker</a></li> (Experimental)
<li><a href="src/content/insertable-streams/webgpu">Integrations with WebGPU for custom video rendering:</a></li> (Experimental)
</ul>
</section>
</div>
<script src="src/js/lib/ga.js"></script>
</body>
</html>